With development of communications technologies, users are requiring increasingly higher quality in voice calls, and a main method for improving voice call quality is increasing bandwidth of a voice signal. If a conventional coding scheme is used for encoding to increase bandwidth of a voice signal, bit rates would be greatly increased. However, a higher bit rate requires larger network bandwidth to transmit the voice signal. Due to constrains of network bandwidth, it is difficult to put into practice a method that increases voice signal bandwidth by increasing a bit rate.
Currently, in order to encode a voice signal with wider bandwidth when a bit rate is unchanged or only changes slightly, bandwidth extension technologies are mainly used. Bandwidth extension technologies include a time domain bandwidth extension technology and a frequency domain bandwidth extension technology. In addition, in a process of transmitting a voice signal, a packet loss rate is a key factor that affects quality of the voice signal. Therefore, how to recover a lost frame as correctly as possible when a packet loss occurs, to make signal transition more natural and more stable when a frame loss occurs is an important technology of voice signal transmission.
However, when a bandwidth extension technology is used, if a frame loss occurs in a voice signal, existing lost frame recovery methods may cause discontinuous transition between a recovered lost frame and frames before and after the recovered lost frame, which causes noise in the voice signal.